voice over IP Seminar Report
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VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP).
In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service.
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VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service
VoIP is therefore telephony using a packet based network instead of the PSTN (circuit switched).
During the early 90's the Internet was beginning its commercial spread. The Internet Protocol (IP), part of the TCP/IP suite (developed by the U.S. Department of Defense to link dissimilar computers across many kinds of data networks) seemed to have the necessary qualities to become the successor of the PSTN.
The first VoIP application was introduced in 1995 - an "Internet Phone". An Israeli company by the name of "VocalTec" was the one developing this application. The application was designed to run on a basic PC. The idea was to compress the voice signal and translate it into IP packets for transmission over the Internet. This "first generation" VoIP application suffered from delays (due to congestion), disconnection, low quality (both due to lost and out of order packets) and incompatibility.
VocalTec's Internet phone was a significant breakthrough, although the application's many problems prevented it from becoming a popular product. Since this step IP telephony has developed rapidly. The most significant development is gateways that act as an interface between IP and PSTN networks.

What is Voice Over IP?

Voice over IP (VoIP) is a blanket description for any service that delivers standard voice telephone services over Internet Protocol (IP). Computers to transfer data and files between computers normally use Internet protocol.
"Voice over IP is the technology of digitizing sound, compressing it, breaking it up into data packets, and sending it over an IP (internet protocol) network where it is reassembled, decompressed, and converted back into an analog wave form.." The transmission of sound over a packet switched network in this manner is an order of magnitude more efficient than the transmission of sound over a circuit switched network.
As mentioned before, VoIP saves bandwidth also by sending only the conversation data and not sending the silence periods. This is a considerable saving because generally only one person talks at a time while the other is listening. By removing the VoIP packets containing silence from the overall VoIP traffic we can reach up to 50% saving. In a circuit switched network, one call consumes the entire circuit. That circuit can only carry one call at a time.

In a packet switched network, digital data is chopped up into packets, sent across the network, and reassembled at the destination. This type of circuit can accommodate many transmissions at the same time because each packet only takes up what bandwidth that is necessary.. Internet Telephony simply takes advantage of the efficiencies of packet switched networks.
Gateways are the key component required to facilitate IP Telephony.
A gateway is used to bridge the traditional circuit switched PSTN with the packet switched Internet.
The gateway allows the calls to transfer from one network to the other by converting the incoming signal into the type of signal required by the network it is required to send it on. For example, A PC user wishes to call someone using a conventional phone. The PC sends the IP packets containing digitized voice to the gateway.

Requirements of a VoIP

The requirements for implementing an IP Telephony solution to support Voice Over IP varies from organization to organization, and depends on the vendor and product chosen. The following section aims to identify the fundamental requirements in the general case and is split into 3 sections:

Software Requirements
Hardware Requirements
Protocol Requirements

Software Requirements

The software package chosen will reflect the organizational needs, but should contain the following modules as defined in the Technology Guide Series - Voice Over IP Publication, and other sources.

Voice Processing Module. This aspect of the software is required to prepare voice samples for transmission. The functionality provided by the voice processing module should support:

A PCM Interface is required to receive samples from the telephony interface (e.g. a voice card) and forward them to the Voice Over IP software for further processing.

Echo Cancellation is required to reduce or eliminate the echo introduced as a result of the round trip exceeding 50 milliseconds.

Idle Noise Detection is required to suppress packet transmission on the network when there are no voice signals to be sent. This helps to reduce network traffic as up to 60% of voice calls are silence and there is no point in sending silence.

A Tone Detector is required to discriminate between voice and fax signals by detecting DTMF (Dial Tone Multi frequency) signals.

The Packet Voice Protocol is required to encapsulate compressed voice and fax data for transmission over the network.

A Voice Playback Module is required at the destination to buffer the incoming packets before they are sent to the Codec for decompression.

Call Signaling Module. This is required to serve as a signaling gateway which allows calls to be established over a packet switched network as opposed to a circuit switched network (PSTN for example).

Packet Processing Module. This module is required to process the voice and signaling packets ready for transmission on the IP based network.

Network Management Protocol. Allows for fault, accounting and configuration management to be performed.

Hardware Requirements

The exact hardware, which would be required, again, depends on organizational needs and budget. The list below highlights the most general hardware required.
The most obvious requirement is the existence (or installation) of an IP based network within the branch office gateway is required to bridge the differences between the protocols used on an IP based network and the protocols used on the PSTN.
The gateway takes a standard telephone signal and digitizes it before compressing it using a Codec. The compressed data is put into IP packets and these packets are routed over the network to the intended destination.
The PC's attached to the IP based network require the voice/fax software outlined above. They also require Full Duplex Voice Cards which allow both communicating parties to speak at the same time - as often happens in reality.
As an alternative to installing Voice Cards, IP Telephones can be attached to the network to facilitate Voice Over IP. A secondary gateway should be considered as a backup in the event of the failure of the primary gateway.

Protocol Requirements

There are many protocols in existence but the main ones are considered to be the following:

H.323 is an ITU (International Telecommunications Union) approved standard which defines how audio /visual conferencing data is transmitted across a network. H.323 relies on the RTP (Real-Time Transport Protocol) and RTCP (Real Time Control Protocol) on top of UDP (User Datagram Protocol) to deliver audio streams across packet based networks.

G.723.1 defines how an audio signal with a bandwidth of 3.4KHz should be encoded for transmission at data rates of 5.3Kbps and 6.4Kbps. G.723.1 requires a very low transmission rate and delivers near carrier class quality. The VoIP Forum as the baseline Codec for low bit rate IP Telephony has chosen this encoding technique.

G.711. The ITU standardised PCM (Pulse Code Modulation) as G.711. This allows carrier class quality audio signals to be encoded for transmission at data rates of 56Kbps or 64Kbps. G.711 uses A-Law or Mu-Law for amplitude compression and is the baseline requirement for most ITU multimedia communications standards.

Real-Time Transport Protocol (RTP) is the standard protocol for streaming applications developed within the IETF (Internet Engineering Task Force).
Resource Reservation Protocol (RSVP) is the protocol which supports the reservation of resources across an IP network. RSVP can be used to indicate the nature of the packet streams that a node is prepared to receive.

How VoIP works

How VoIP works : Part 1

Let us look at very simple VoIP call. Consider two VoIP telephones connected via an IP network .In this example both VoIP telephones are connected to a local LAN. Sallyâ„¢s phone has an IP address of ,Billâ„¢s phone is, the IP addresses uniquely identify the telephones. Both our phones are configured to use a widely used VoIP standard called H.323.
Bill wants to talk to Sally and his phone knows the IP address of Sallyâ„¢s phone. Bill lifts the handset and 'dials' Sally, the phone sends a call setup request packet to Sally's phone, Sallyâ„¢s phone starts to ring, and responds to Bill's phone with a call proceeding message. When Sally lifts the handset the phone sends a connect message to Bill's phone. The two phones will now exchange the data packets containing the speech. At the end of the call Bill replaces his handset and phone stops sending voice data sends a disconnect message and Sally's phone responds with a release message. The call is now complete. all the messages contain the Q931 ISDN protocol.
Having introduced VoIP I will now talk about three main 'types' of VoIP installed in the market place today.

Main Ëœtypesâ„¢ VoIP

VoIP has broadly three main branches, which can and do overlap.

VoIP over the Internet This is probably the best known and most publicized, talking PC to PC. Basically free telephone calls. The call is only free if both parties to the call have access to the public Internet at zero cost..
Advantage... free calls regardless of distance or length of call.
Disadvantage.... often the voice quality is bad due to the lack of bandwidth available for the call.
Other factors. Have to use a PC or other computer running VoIP software.

Office to Office A large multinational company will have offices across the whole country. They have a fixed data network connecting all the offices together. This allows every computer access to every other computer in the company. By installing a VoIP Gateway in each office and connecting it to the office legacy PBX and to the data network, employees use the data network for voice calls between offices.

Interoffice calls are free, since the company already has the bandwidth between offices. The technology is transparent to the user, and requires minimum training. The only new equipment required is a gateway at each office. Voice quality is good, because the company has control over the bandwidth.

Extra bandwidth may be required between offices, which offset the savings.
Other factors... The carrier providing the interoffice bandwidth will almost certainly offer an alternative solution including management of the internal telephone traffic.

IP PBX A traditional Private Branch Exchange (PBX) connects all the phones within an organization to the public telephone network. Essentially IP PBX replaces all the internal phones with VoIP telephones. The IP PBX has standard telephone trunk connections to the public telephone network. The IP PBX is a PBX with VoIP, but it also

has the ability to support VoIP over the Internet and Office to Office VoIP.
Advantages. Single cable infrastructure. The technology is transparent to the user, and requires minimum training. Future proof technology.
Disadvantages. Primarily useful for Greenfield sites, but can be adapted to work with existing technology.

How VoIP works part 2 : The Protocols.

I have made an assumption that both ends of a VoIP telephone conversation are compatible. This compatibility only happens if both ends agree to use the same protocol. All manufacturers who claim to be producing industry standard voice over IP either support SIP or H.323 protocol.

So what is H.323 ?

Over the next few years, the industry will address the bandwidth limitations by upgrading the Internet backbone to asynchronous transfer mode (ATM), the switching fabric designed to handle voice, data, and video traffic. Such network optimization will go a long way toward eliminating network congestion and the associated packet loss. The Internet industry also is tackling the problems of network reliability and sound quality on the Internet through the gradual adoption of standards. Standards-setting efforts are focusing on the three central elements of Internet telephony: the audio codec format; transport protocols; and directory services.

H.323 Call Sequence :

As such, H.323 addresses the core Internet-telephony applications by defining how delay-sensitive traffic, (i.e., voice and video), gets priority transport to ensure real-time communications service over the Internet. (The H.324 specification defines the transport of voice, data, and video over regular telephony networks, while H.320 defines the protocols for transporting voice, data, and video over integrated services digital network (ISDN).

How VoIP works part 3: Encoding

The call control part of H.323 sets up the parameters for the full duplex voice path between source telephone and destination telephone. I will continue with my analogies to explain how your voice gets transported across the Internet.
In terms of H.323 there is a trade off between call quality and bandwidth, in general the higher the quality the greater the bandwidth required
During the call setup portion of H.323 the phones have to decide which speech encoder/decoder to use when they send the speech to the other phone, Bill and Sally both have phones that support G.723.1, G.711 and G729.
The main difference between each of these encoders is the amount of bandwidth they use, G.711 uses 64kbit/s and G.723.1 can use as little as 5.3kbit/s. Although it would seem obvious to use the encoder with the lowest bandwidth, there is a loss of quality with a lower bandwidth.. At the same time a stream of G723.1 encoded voice data starts being sent from each phone to the other phone.
How VoIP works part 4 :Hear the Quality.
The performance of the speech encoders at each end, the number of packets lost on route, Latency and Jitter.
I have already talked about the encoders in the previous section. I also bundle into the encoding process echo suppression. In the early days of voice calls via satellite there would be an annoying echo. As the technology improved the echo disappeared. Echo suppression is very key to good quality VoIP calls . I do not dwell on the subject since the mathematics is beyond my comprehension. Good echo suppression makes for quality calls.
Be warned that because a manufacturer has a G.723.1 encoder it may not sound the same as another manufacturer who claims to have G.723.1, quality does vary. As a general rule the occasional lost packet will not affect too drastically the quality of a call, but lose 5 in a row and an entire word is lost and this will be a problem. So if you are going to have lost packets make sure they are only lost in a regular distributed manner. 5% lost packets distributed evenly will not result in the loss of words lose 5% of the words by clustering the packets and the effect is bad.

Advantages of VoIP

There are many advantages to be gained from implementing an IP Telephony solution within the organization. The following list aims to highlight some of the advantages of such a strategy:
Single network infrastructure. When installing VoIP in the office only a single cable is required to the desk, for both telephone and data. Eliminating separate telephone wiring.
VoIP uses "soft" switching which eliminates most of the legacy PBX equipment. Reducing the cost of installing a communications infra-structure and the maintenance cost once installed.
Simple upgrade path. The VoIP PBX technology is software based. It is easier to expand, upgrade and maintain than its traditional telephony counterparts.

Bandwidth efficiency. VoIP can compress more voice calls into available bandwidth than legacy telephony.. IP Telephony helps to eliminate wasted bandwidth by not transporting the 60% of normal speech which is silence

IP - the underlying protocol - is supported by most platforms and is independent of the transport protocol used.

Only one physical network is required to deal with both voice/fax and data traffic instead of two physical networks. Having only one physical network has the following advantages:

lower physical equipment cost ,lower maintenance costs.


While there are many aspects of VoIP which provide considerable benefits, the technology is still very young and problems remain. The following section looks at some of the weaknesses of this technology and their consequences.
The Internet is not the best medium for real time communications. Individual packets can take different routes and varying delays can be encountered and packets lost in transit. Waiting for delayed packets or retransmission of lost packets can result in considerable degradation of quality. Long delays in transit can affect quality so much that the technology can become unusable, though many vendors do have solutions which aim to negate the degradation suffered due to transit delays.
While some standards have been set by the ITU, the technology is not fully standardized and there is no guarantee that products from different vendors will be interoperable. Some vendors are trying to resolve this problem by forming groups and making guarantees about the products in the group but this is only a partial solution - vendors outwith the group cannot guarantee interoperability.
Heavy congestion on the network can result in considerable degradation of service as IP is not good at providing QoS (Quality of Service) guarantees. Feedback to Lucent Technologies customers reflect this worry. Major companies are planning to install IP Telephony capabilities at some point and have carried out initial investigations, however:
Since only one physical network for both data and voice/fax transmissions is required, failure of the network could be catastrophic, as all communications capabilities are lost.


Many vendors offer the ability to incorporate Virtual Private Networking (VPN) with relative ease into the IP Telephony solutions they provide. This allows any transmission to be encrypted using a number of cryptographic techniques and providing security by transmitting the communications through a 'tunnel' which is set up using PPTP (Point-to-Point Tunneling Protocol) before commencing communications.

IP Telephony allows companies to exploit Computer Telephony Integration to its full extent.
The convergence of communications technologies allows greater control over communications, most vendors provide logging and accounting facilities whereby all usage can be monitored.

Conclusion :

Without a doubt, the data revolution will only gain momentum in the coming years, with more and more voice traffic moving onto data networks. Vendors of voice equipment will continue to develop integrated voice and data devices based on packetized technology. Users with ubiquitous voice and data service integrated over one universal infrastructure will benefit from true, seamless, transparent interworking between voice and all types of data.


1. Computer Networks by Andrew S.Tanenbaum
2. Internetworking with TCP/IP by Douglas E.comer
3. http://www.iec.org.com
4. http://www.telogy.com
5. http://www.rad.com
6. http://seminar and presentationproject and implimentations.com

The Voice-Over Internet Protocol (VoIP) technology allows the voice information to pass over IP data networks. This technology results in huge savings on the amount of physical resources required to communicate by voice over long distance. It does so by exchanging the information in packets over a data network.

The basic functions performed by a VoIP include “ signalling, data basing, call connect and disconnect, and coding/decoding. The steps involved in originating and internet telephone call are the conversion of the analogue voice signal to digital format and compression/translation of the signal into internet protocol (IP) packets for transmission over the internet; the process is reversed at the receiving end. VoIP software™s like Vocal TEC or Net 2 Phone are available for the user. With the exception of phone to phone, the user must posses an array of equipment which should at minimum include VoIP software, an internet connection, and a multimedia computer with a sound card, speakers, a microphone and a modem.

The VoIP network acts as a gateway to the existing PSTN network. This gateway forms the interface for transportation of the voice content over the IP networks. Gateways are responsible for all call origination, call detection, analogue to digital conversion of voice, and creation of voice packets.
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18-05-2010, 10:56 AM

VoIP, the latest technology that ensures you to have a better clarity of the voice is bound to be a huge hit among the people, but the problem is the cost effectiveness at the earlier stages. As the wiring for this is quite complex and the old ones need to be reinstalled to get started with the process. This could be the best example of the emerging trend of the technology and the way things would be changing in the coming years. Thanks for providing a brief information about VoIP, got to know a little more about it.
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.ppt   Voice Over Internet Protocol(VoIP).ppt (Size: 63.5 KB / Downloads: 251)
What is Voice over Internet Protocol?

Conversion of sound into digital signal
Encapsulation in IP packet
Transport over network
De-encapsulation and reconversion of digital signal into sound

This type of voice over IP has been around a long time.

Seldom used because the cost of long distance has dropped dramatically – domestic at least
Reliability and quality of transmission not high enough for business communications (or even personal use)
Difficulty in set up and use

The basic components of VoIP are:

Voice Processing
Call Processing
Packet Processing
Network Management

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voice over IP Seminar Report


VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service
VoIP is therefore telephony using a packet based network instead of the PSTN (circuit switched).
During the early 90's the Internet was beginning its commercial spread. The Internet Protocol (IP), part of the TCP/IP suite (developed by the U.S. Department of Defense to link dissimilar computers across
many kinds of data networks) seemed to have the necessary qualities to become the successor of the PSTN.
The first VoIP application was introduced in 1995 - an "Internet Phone". An Israeli company by the name of "VocalTec" was the one developing this application. The application was designed to run on a basic PC. The idea was to compress the voice signal and translate it into IP packets for transmission over the Internet. This "first generation" VoIP application suffered from delays (due to congestion), disconnection, low quality (both due to lost and out of order packets) and incompatibility .Vocal Tec’s Internet phone was a significant breakthrough, although the application's many problems prevented it from becoming a popular product. Since this step IP telephony has developed rapidly. The most significant development is gateways that act as an interface between IP and PSTN network

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Voice over Internet Protocol


This article needs attention from an expert on the subject. See the talk page for details. WikiProject Law or the Law Portal may be able to help recruit an expert. (November 2008)

Voice over Internet Protocol (VoIP, Voice over IP) is a general term for a family of methodologies, communication protocols, and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms frequently encountered and synonymous with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.

Internet telephony refers to communications services — voice, facsimile, and/or voice-messaging applications — that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.[1]

VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. Codec use is varied between different implementations of VoIP (and often a range of codecs are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs.
Voice over IP has been implemented in various ways using both proprietary and open protocols and standards. Examples of technologies used to implement Voice over IP include:
• H.323
• IP Multimedia Subsystem (IMS)
• Media Gateway Control Protocol (MGCP)
• Session Initiation Protocol (SIP)
• Real-time Transport Protocol (RTP)
• Session Description Protocol (SDP)
The H.323 protocol was the one of the first VoIP protocols that found wide-spread implementation for long-distance traffic, as well as local area network services. However, since the development of newer, less complex protocols, such as MGCP and SIP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic. In particular, the Session Initiation Protocol (SIP) has gained widespread VoIP market penetration.
A notable proprietary implementation is the Skype protocol, which is in-part based on the principles of peer-to-peer networking.

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.pptx   voip-1226295676997160-9.pptx (Size: 424.44 KB / Downloads: 87) VOICE OVER INTERNET PROTOCOL


The Definition

Voice Over Internet Protocol:
A technology that allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line.

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What is VoIP?
“Technology used to transmit voice conversations over a data network using the Internet Protocol. Such data network may be the Internet or a corporate Intranet.” (http://www.shiftnetworks.com/glossary.cfm)


“Ability of an IP network to carry telephone voice signals as IP packets in compliance with International Telecommunications Union Telecommunication Standardization Sector (ITU-T) specification H.323. VoIP enables a router to transmit telephone calls and faxes over the Internet with no loss in functionality, reliability, or voice quality.” (http://www.trapezenetworks.com/technolog...sary_6.asp)

“VoIP works through sending voice information in digital form in packets, rather than in the traditional circuit-committed protocols of the public switched telephone network.” (http://www.interconnect.co.za/links.html)


“Voice over IP (also called VoIP, IP Telephony, and Internet telephony) refers to technology that enables routing of voice conversations over the Internet or any other IP network. The voice data flows over a general-purpose packet-switched network, instead of the traditional dedicated, circuit-switched voice transmission lines. (en.wikipedia.org/wiki/VoIP)

How is this Useful?

“VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you are bypassing the phone company (and its charges) entirely.”



VoIP as Revolutionary Technology

“It has the potential to completely rework the world's phone systems, because it uses packet-switching rather than circuit-switching. Which makes VoIP more efficient, and less costly.”

Three different Flavors of VoIP
Three different ways to place a call are;
Simplest and most common way
Uses a device called an ATA (analog telephone adaptor). Which allows us to connect a standard phone to our computer or our Internet connection for use with VoIP. The ATA is an analog-to-digital converter.
IP Phones
Specialized phones look just like normal phones with a handset, cradle and buttons.
Uses an RJ-45 Ethernet connector instead of standard RJ-11 phone connectors.
Connect directly to our router and have all the hardware and software necessary right onboard to handle the IP call.
Easiest way to use VoIP.
All we need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like DSL modem.

How stuff Works?

Existing phone systems are driven by a very reliable but somewhat
inefficient method for connecting calls called circuit switching.
- When a call is made between two parties, the connection is
maintained for the duration of the call.
- Because we’re connecting two points in both directions, the
connection is called a circuit. This is the foundation of the Public
Switched Telephone Network (PSTN).

Here's how a typical telephone call works:
We pick up the receiver and listen for a dial tone. This let us know that we have a connection to the local office of our telephone carrier.
We dial the number of the party we wish to talk to.
The call is routed through the switch at our local carrier to the party we are calling.
A connection is made between our telephone and the other party's line using several interconnected switches along the way.
The phone at the other end rings, and someone answers the call.
The connection opens the circuit.
We talk for a period of time and then hang up the receiver.
When we hang up, the circuit is closed, freeing our line and all the lines in between.
Circuit switching keeps the connection open and constant.
VoIP uses Packet switching for transmitting voice information into digital form.
Packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet, from one system to another.
It works like this:
The sending computer chops data into small packets, with an address on each one telling the network devices where to send them.
Inside of each packet is a payload (a piece of the e-mail, a music file or whatever type of file is being transmitted inside the packet).
The sending computer sends the packet to a nearby router and forgets about it. The nearby router send the packet to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router, and so on.
When the receiving computer finally gets the packets (which may have all taken completely different paths to get there), it uses instructions contained within the packets to reassemble the data into its original state.
Packet switching is very efficient because it lets the network route the packets along the least congested and cheapest lines. Also frees up the two computers communicating with each other so that they can accept information from other computers, as well.
How it is that analog audio is turned into packets for VoIP transmission?

Stands for coder-decoder.
Converts an audio signal into a compressed digital form for transmission and then back into an uncompressed audio signal for replay.
Accomplish the conversion by sampling the audio signal several thousand times per second.
Convert each tiny sample into digitized data and compresses it for transmission.
A G.729A codec has a sampling rate of 8,000 times per second.
Operate by using advanced algorithms that help them sample, sort,
compress and packetize audio data.
Codec works with the algorithm to convert and sort everything out, but none of that is any good without knowing where to send the data?

Soft Switches:
E.164 is the name given to the standard for the North American Numbering Plan (NANP).
Which is numbering system that phone networks use to know where to route a call based on the numbers entered into the phone keypad. In that way, a phone number is like an address:
(313) 555-1212 313 = State
555 = City
1212 = Street address
To route the phone call to the region denoted by the area code.
The "555" prefix sends the call to a central office.
Network routes the call using the last four digits. which are associated with a specific location. So based on that system, no matter where you are in the world, the number combination "(313) 555" will always put you in the same central office, which has a switch that knows which phone is associated with "1212."
Soft Switches: (Contd.)
The central call processor is a piece of hardware running a specialized database/mapping program called a soft switch. Think of the user and the phone or computer associated with that user as one package -- man and machine. That package is called the endpoint. The soft switch connects endpoints.
Soft switches know:
Where the endpoint is on the network
What phone number is associated with that endpoint
The current IP address assigned to that endpoint
So when a call is placed using VoIP, a request is sent to the soft switch asking which endpoint is associated with the dialed phone number and what that endpoint's current IP address is. The soft switch contains a database of users and phone numbers. If it doesn't have the information it needs, it hands off the request downstream to other soft switches until it finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests. It sends back all the relevant information to the softphone or IP phone, allowing the exchange of data between the two endpoints.
How do these different pieces of hardware and software will communicate efficient and will understand each other?
Several protocols are currently used for VoIP.
They define ways in which devices like codecs connect to each other and to the network using VoIP.
They also include specifications for audio codecs.
The most widely used protocol is H.323, a standard created by the International Telecommunication Union (ITU).
H.323 is a comprehensive and very complex protocol that was originally designed for video conferencing. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP.
H.323 is a suite of protocols and also incorporates many individual protocols that have been developed for specific applications.
An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP). SIP is a much more streamlined protocol, developed specifically for VoIP applications. Smaller and more efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the process

Media Gateway Control Protocol (MGCP) is a third commonly used VoIP protocol that focuses on endpoint control. MGCP is geared toward features like call waiting.

Frequently Asked Questions (FAQs)
If I have VoIP service, who can I call? Depending upon your service, you might be limited only to other subscribers to the service, or you may be able to call anyone who has a telephone number - including local, long distance, mobile, and international numbers. If you are calling someone who has a regular analog phone, that person does not need any special equipment to talk to you. Some VoIP services may allow you to speak with more than one person at a time.
Can I use my Computer While I talk on the Phone? In most cases, yes. Does my Computer Have to be Turned on? Only if your service requires you to make calls using your computer. All VoIP services require your broadband Internet connection to be active.
How Do I Know If I have a VoIP phone Call? If you have a special VoIP phone or a regular telephone connected to a VoIP adapter, the phone will ring like a traditional telephone. If your VoIP service requires you to make calls using your computer, the software supplied by your service provider will alert you when you have an incoming call.

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.ppt   ppt1.ppt (Size: 127.5 KB / Downloads: 70)
Voice Over IP” ?
 It is a technology that allows to make telephone calls using Data Network (Internet) to other people using same VoIP service, other VoIP service, or a telephone
no. including local, long distance, ISD and
How It Works ?
 VoIP converts the voice signal from the telephone into a digital signal that travels over the Internet then converts it back at the other end so you can speak to anyone with a regular phone number
Types of VoIP Services
 PC-To-PC Calling
 Hotline
 Card Calling
VoIP …
 Types of VoIP Adoption
 Dedicated
 Hosted
 Equipment Required
 Broadband Connection
 Computer/Telephone
 Cable modem/Telephone Adapter
 VoIP Router(VoIP Gateway)
 MicroPhones, Speakers
 VoIP …
 Some VoIP providers are Vonage, Skype,
Cisco, Avaya, VoicePulse etc.
Enhanced Features

 Click-to-call
 Web and Audio Confrencing
 Find me/Follow me
 Local Number portablity
 Unified Messaging
 Auto call Distribution
Advantages …
 Streamlined Operations
 4-5 Digit dialing to network of offices
 Unified auto attendant
 Unified office attendant portal
 Reduced costs
 Lower usage cost
 Lower maintenance
 Reduced network infrastructure
 Business continuity
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.doc   Voice over internet protocol (VoIP).doc (Size: 79 KB / Downloads: 52)
The main objective of this paper is to provide a conceptual view on “Voice over Internet Protocol” (VoIP), its techniques characteristics and functions. What’s Voice on Internet Protocol? The benefits, the VoIP market, VoIP solutions, and implementation. It is the most important concept in IT and as well as in Business point of view.
This brochure is for small businesses who want to cut the cost of their phone bill. It’s also for those who want to integrate their phone system with their business applications, to make staff more productive and offer better customer service and it covers the benefits of implementing VoIP, with advice on how to go about it.
 What is VoIP?

Many small businesses are faced with spiraling phone bills. One way of reducing your bills is to use your IT network for phone calls between your branches, and to use the internet for external calls. This is known as Voice over Internet Protocol (VoIP).
VoIP can also dramatically improve the way you work. If you switch phone calls to IT network, you can merge voice and data, boosting staff productivity and enhancing customer service. As the price of high speed internet Connections comes down, it’s well worth taking a fresh look at long term plans for phone and IT network. Protocols are the set of rules that should govern data communications.
VoIP enables businesses to make phone calls across computer networks, providing a low cost and efficient way to complement traditional phone systems. VoIP can be used in local office networks or between sites, enabling you to integrate call handling with other parts of your business such as your website.
The benefits:
1) The main advantage of VoIP is cheaper phone calls. Another key advantage is being able to combine phone calls with business data. It means you can adopt call centre style technology, with each incoming call triggering onscreen pop ups with customer details.
2) As the cost of high speed internet access (broadband) comes down, VoIP is now within reach of small businesses. Some telecoms companies and internet service providers are now offering VoIP deals targeted at the small business sector.
3) With VoIP, voice data is compressed and transmitted over a computer network. This means VoIP uses up to 90% less bandwidth than a traditional phone call and is consequently more cost-effective and more efficient.
4) VoIP is helping the phone companies save money, and by introducing a VoIP phone system on your own computer networks, you could do so too.
For any business, the immediate benefits can be:
• Cheaper external, long distance and international calls for the price of a local call.
• Free internal calls to all parts of your company that share a computer network. With
a Virtual Private Network (VPN) in place you can speak to connected colleagues at
different branches or on the road free of charge.
5) Cheaper calls are not only advantage.If you merge phone and networks, VoIP helps you work more efficiently and make company’s phone network easier to look after.
• Simpler infrastructure: With VoIP on your computer network you can add
telephones and increase call capacity without running additional cabling.
• Scalability: Traditional PBX (Private Branch Exchange) phone systems have a set
number of ports for telephones to plug in to.
• Reduce operating costs: Because a VoIP-enabled system is based on software
rather than hardware, it is easier to manage and maintain.
• Improve productivity: VoIP treats voice as if it were any other kind of data, so
users can attach documents to voice messages or participate in virtual meetings
using shared data and Videoconferencing.
• Wireless-compatible: With a wireless LAN in place, mobile devices like PDAs and
smart phones can use your VoIP system. (If you install a wireless LAN, you need
to make sure you have appropriate security measures in place.)
• Enhanced customer service: A VoIP enterprise can put web users in touch with customer service staff on website. You could also look at implementing customer relationship management software (CRM).
• Dependable call management: Voice-related services, such as follow me
roaming, caller-ID, call forwarding and broadcast messaging, become simpler to
maintain and can be updated as needed by your employees.
• Flexibility: VoIP systems provide greater flexibility as you can run a number of ‘Virtual users’ through each network socket.
The VoIP market:
The good news is that the UK communications industries regulator, is pushing suppliers to introduce low-cost voice over broadband services in the UK. These services could be of interest to small and medium-sized companies.
As it’s a competitive market to see a raft of new products and services over the next few years, from traditional telephone service suppliers, Internet Service Providers (ISPs) and network suppliers. Managed services where phone and data network are likely to be popular options for those who don’t have in-house expertise.
VoIP solutions getting started and necessities:
VoIP allows phone calls to be made between PCs that are connected on a computer network. This can be an internal LAN (Ethernet, wireless-based) or any computer connected to the internet.
To make and receive VoIP calls internally in your business you will each need a multimedia equipped PC (Pentium) or an Apple G4.
You also need to be connected to some kind of network:
• Internet: Home users who have an internet connection you can, phone any suitably
equipped PC in the world free of charge with a dial-up connection. But for serious
business use, you need a high speed internet connection such as broadband.
• Private networks: VoIP can work across almost any data network, including
wireless or Ethernet-based LANs and Virtual Private Networks (VPN), as well as the
internet itself. The quality of service depends on congestion and transmission speeds
of the network in question.
You need appropriate software to make VoIP calls on PC or Apple Mac. The latest operating systems include applications that let you make PC-to-PC calls.
• Microsoft Windows 2000 users can use Net meeting.
• In Windows XP, Messenger has this capability.
• Apple users use iChat which comes with Mac for use with broadband connections.
• Applications such as Skype let PC, Mac and Linux users chat free of charge over
the internet. The software is suitable for personal use rather than business use.
You could use computer’s built-in microphone and sound card to make and receive calls, but most users find headsets and handsets more practical.
• Analogue handsets plug into your existing soundcard. They are simple to operate
but sound quality can be variable.
• USB handsets plug into any PC and deliver audio as they have built-in sound cards.
SIP models can plug into any PC and make/receive calls from anywhere in the world.
• Analogue telephone adaptors units that convert your existing analogue phone
into an internet capable (SIP) phone.
• IP phones are available, which plug into the data network. Making phone calls over
the internet, you can check email, browse the internet and access your company’s
business applications.
To call people who have phones rather than VoIP-enabled PCs, your company needs a modern, IP-enabled PBX (private branch exchange). They come with a standard gateway which connects VoIP calls to the public phone network.
You may need to download the service provider’s software to convert your PC into a telephone, and use a PIN number to access the service. Using an Internet Telephony Service Provider is a low cost option, as you do not need to invest in additional hardware.
PC-to-phone services for businesses can be set up with single or multiple user accounts (with itemized billing) and can be a cost effective solution if you need to make international or long distance calls on a regular basis.
If reducing your international phone bill is the sole concern you can even opt for an account with an Internet Telephony Service Provider that gives you access to its low tariffs via a calling card solution. Here, you subscribe to a VoIP service and pre-dial its code when you are calling abroad. You don’t have to upgrade your network to take advantage of these services, so it’s a low-cost option.
VoIP solutions going further
VoIP is not just about cheaper phone calls. If you merge your phone and data network, it can dramatically change the way you work, improving customer service along the way. For example, you can use call centre technology to improve the way you deal with customers. Each incoming call can trigger your database to show a customer’s address details and transaction history on screen.
So if you’re planning to upgrade your telephone system or if you’re implementing broadband on your IT network, it would be wise to make sure any new systems you install can cope with VoIP.
Most businesses use Private Branch Exchanges (PBXs) to manage shared external lines and switch calls between users on internal lines.
By taking this route, businesses can preserve their existing investment, and take advantage of VoIP. You may find that the new equipment pays for itself within a year, through the costs you save on your phone bill.
To do this you will need:
• An IP-Private Branch Exchange server (IP-PBX server). It deals with call routing
and connection requests, monitors data traffic and manage bandwidth allocation.
• A gateway which provides the bridge between VoIP traffic and the standard
telephone network.
• Software that allows multimedia-capable PCs to operate as high performance
telephones using the company network
If you are mainly interested in cutting the cost of internal calls between different branch locations, you can install VoIP gateways at each branch’s PBX telephone system and bypass the public telephone system. There is no need for equipment changes for the users, as individual phones are unaffected.
This will create a single, multi-location ‘office’. You will cut the cost of inter-office phone costs and staff will benefit from remotely accessible voicemail and Direct Dial-In (DDI) numbers. The return on investment (ROI) here is straightforward: it’s the reduction in your phone bill less the cost of the hardware. With the ratio of internal to external calls at around 4:1 for the average business, the telephone savings could be substantial.
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.pdf   VoIPpdf.pdf (Size: 100.43 KB / Downloads: 81)
Today, Internet is becoming increasingly popular in usage. It is used not
only for text based data communication only, but also for multimedia
communication, for example, audio, video and graphics. This seminar and presentation deals with
the audio communication over Internet : Voice Over IP. VoIP is the ability to make
telephone calls and send faxes over IP-based data networks with a suitable quality
of service (QoS) and superior cost/benefit . Everyone is talking about VoIP and
everyone wants a piece of the pie.
Voice Over IP (VoIP) enables user to make calls in real time between
any combination of phones and PCs. VoIP converts voice traffic into data packets
and transports them over the Internet. This report first discusses the basic of VoIP
including its motivations and working. Then it focuses on Quality of Services in
VoIP. The protocols and standards that exist today are also examined. Finally the
key problems of present VoIP technology are discussed followed by its expected
Voice Over IP is growing daily and catching on with thousands of
business as well as PC users who have long on this developing technology and its
Available at http://www.mindstien.net
Voice over IP is mainly concerned with the realization of telephone
service over IP-based networks such as the Internet and Intranet. IP telephony is
currently breaking through to become one of the most important service on the net.
The actual breakthrough was made possible by the high bandwidth available in an
intranet and, increasingly, on the internet. Another fundamental reason is the cost
associated with the various implementations.
The public telephone network and the equipment makes it possible are
taken for granted in most parts of the world. Availability of a telephone and access
to low-cost, high quality worldwide network is considered to be essential in
modern society (telephone are even expected to work when the power off).There is,
however, a paradigm shift beginning to occur since more and more communication
is in digital form and transported via packet networks such as IP, ATM cells, and
Frame Relay frames. Since data traffic, there has been considerable interest in
transporting voice over data networks.
Support for voice communications using the Internet Protocol(IP), which
is usually just called “Voice over IP” or VOIP, has become especially attractive
given the low-cost, flat-rate pricing of the public Internet. In fact, toll quality
telephony over IP has now become one of the key steps leading to the convergence
of the voice, video, and data communications industries. The feasibility of carrying
voice and signaling message over the internet has already been demonstrated but
delivering high-quality commercial products, establishing public services, and
convincing users to buy into the vision are just
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